There are a number of attributes that affect perceived voice quality in Voice over IP (VoIP) networks. These attributes include latency and packet loss within the IP backbone network, and choice of voice coder/decoder (codec) at the gateway to the IP network. To ensure toll-quality voice communication, latency and packet loss in the backbone network must be kept below certain thresholds by managing various components of the network architecture.
The architectural components that need to be managed in the backbone network include the number and locations of routers in the backbone network, the transmission capacity deployed between routers, the quality of service mechanisms and routing algorithms employed in the network, the judicious selection of packet size and the appropriate sizing of jitter buffers. At the same time that these components are managed, it is also important to maximize the utilization of network resources to keep costs under control. Cost-effectively maintaining toll-quality voice communication in a VoIP network is a difficult and multi-faceted problem.
Routers in the communication network are limited in packet per second call processing capacity. With the relatively large size of packets used for data services, this does not become an issue. However, since voice communications use significantly smaller packets than data communications to reduce latency, the number of packets per second that must be routed through the network for a voice call is significantly larger than the number of packets to be routed for a data service with the same bandwidth allocation. As latency is reduced, this problem becomes worse, but can be mitigated by allowing latency to increase in certain circumstances.